1. Field of the Invention
The invention relates to methods and apparatus for improving voice quality in compressed audio signals transmitted over packet networks. More specifically, the invention relates to methods and apparatus for reducing the degradation of the quality of the voice transmission due to repeated compression and decompression.
2. Description of the Background Art
Voice connections are commonly established using packet networks. Once a connection over a packet network is established, signals are sent over the network in packets. The signals sent over the network are digitized from either analog or digital sources. To transmit voice over a packet network, the voice signal is compressed and placed into data packets. These packets are transmitted over a packet network, received at the other end, decompressed and passed on as voice information. Three kinds of packets are normally transmitted over the packet network in a voice connection. These are:
Voice packets. These packets contain compressed voice. Their size varies depending on the codec in use.
SID packets. These packets are transmitted at the beginning of a silence interval. By using these packets, the packet network bandwidth is saved, as compared to what it would cost by sending the silence compressed as voice packets.
DTMF packets. DTMF tones do not compress and reproduce well enough for the telephony equipment to work properly if compressed using some popular low-bit-rate voice codecs. Therefore, these tones are identified and transmitted in a different packet format over the packet network. This also provides the opportunity of moving DTMF signaling to out-of-band signaling, and vice versa.
In certain situations a connection is established between two packet networks over a PBX/PSTN interconnection. In these circumstances the data will travel over PSTN lines as digital PCM samples.
In addition, some signaling packets are also passed that are used for maintaining the connection. DTMF tones, and some other tones, are used by the Central Office PBX for establishing the connection in the beginning of a call. However, once a voice connection has been established, it is generally assumed that the routing equipment (PBX) will not make use of DTMF digits. Therefore, they can be transported over the PCM interface in a packet format.
Since the data may have to travel over PSTN lines as PCM samples, it is important to consider the following properties of PSTN lines:
Bit robbing. In North America, one least significant bit of every sixth frame is robbed and used for signaling in channel associated signaling schemes. This means that one out of six bits will be corrupted. Further, sequential PBX switches will also rob every sixth frame of the LSB. However, the robbed bit will most likely not be the same sixth bit, resulting in two or more bits out of six robbed, increasing the difficulty of identification of any sequence embedded in the LSB.
Frame slips. Frame slips may occur over PSTN lines. If a frame slip occurs, one PCM frame may be either lost, or repeated twice. In the worst case, it can be anticipated that these will occur every 5.6 seconds. This phenomenon may interfere both with signature sequence detection, and packet retrieval/synchronization.
Digital pads. We will assume that there are no digital pads over the link.
Asynchronous tandem T1 links. Asynchronous tandem T1 links may cause more than one out of six LSB of the PCM samples to be corrupted.
The invention is applicable to the situations where two or more codecs appear in a series in a voice connection. Every time voice is encoded and decoded through a low-bit-rate codec, there is degradation in the voice quality. FIG. 1 illustrates a telecommunication connection over a PBX utilizing Golden Gateway software from Telogy Networks, Inc. In FIG. 1, a connection between two terminal pieces of telecommunication equipment 10 and 11 has been established over two packet networks, through a PBX/central office 12 that is remote to both 10 and 11. The equipment 10 and 11 can be telephones, facsimile machines, computers, voice processing equipment or any combination of telecommunication equipment. Once a connection has been established between the equipment 10 and 11, the first gateway 13 uses a first codec to compress voice 13 over the first packet network 14. Then the voice packets are decompressed by gateway 15 and transmitted as PCM samples over the Central Office PBX/PSTN lines 16. The voice signal is received by gateway 17 and compressed again for transmission over the second packet network 18 and where it is received and again decompressed by gateway 19 and provided to second telephone 11.
The use of two codecs in series, tandem codecs, requires repeated compression and decompression. Repeated compression and decompression result in degradation in voice quality. Such a situation may also arise in the presence of call-forwarding over two long-distance links.
The present invention alleviates the degradation in voice quality caused by successive compression and decompression by sending the compressed voice packets on the first packet 14 network directly to the second packet network 15 without decompression and recompression. This form of transmission is referred to herein as tandem transparent mode. In order to accomplish this, the present invention teaches to embed the packet information in compressed form into the voice connection of PCM samples. However, to achieve this result, the gateways 15 and 17 must be able to realize that the other gateway is not terminal equipment (e.g., a telephone, facsimile machine, or the like). Both gateways 15 and 17 must recognize the other as a forwarding gateway that normally compresses/decompresses and forwards the compressed voice packets over a packet network or forwards the voice signal to the terminal equipment. By recognizing the other gateway on the other side of the PBX, each gateway can eliminate decompression and transmit the compressed voice packets as data within the PCM samples (transparent mode). The receiving gateway will recognize the PCM samples as packet data and forward the packets on to the packet network without recompression, achieving improved voice quality. The connecting gateways can only detect the possibility of tandem codecs after the voice connection has been established.